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• How can I make a Transcoded Video Filestream using C# and .NET Core

  25 avril 2021, par Drew Chase

  Overview

  


  I'm currently working on a media streaming server using ASP.net Core REST Server. I'm currently using .net 5.0 and ASP.net Core MVC

  


  What I need

  


  I need to be able to dynamically down-res the original video file. from 1080p to 720p for example.
Also I need to be able to make the media file able to be transcoded to a different encoding based on client capabilities.

  


  What I've Tried

  


  I've been looking for a library that can manage this feat, but I can't seem to find one. I thought FFMpeg would be able to do this. I know this is possible because applications like plex and emby seem to manage this.

  


  What I've Done

  


  [HttpGet("/api/streaming/video")]
public IActionResult GetFile()
{
    string path = "C:\Path\To\Video\FILE.mp4";
    System.IO.FileStream stream = new(path, System.IO.FileMode.Open, System.IO.FileAccess.Read);
    Microsoft.AspNetCore.Mvc.FileStreamResult file = File(stream, "video/mp4", true);
    return file;
}


  


  Framework Tried

  


  

  • Xabe.FFmpeg

  


  • FFMpegSharp

  


  


• Best approach to set up live video streaming between HoloLens2 and Desktop PC

  19 mai 2021, par SilverLife

  First of all, I am sorry for this question. I know its more asking for an advice than asking about any coding problems. Unfortunately I don´t know where else I should ask such a question, so please be patient. (I am open for any recommendations)

  


  Since a while I am searching for a promising approach to stream live videos from a desktop PC via network to my HoloLens 2 unity application. The video transfer should be within a encapsulated network without internet access. Therefore a direct connection between both devices without any signaling- or web-servers would be desirable, if possible. For now we are thinking about sending the stream via ffmpeg and receiving it somehow in the unity application.
So far I came across MixedReality-WebRTC and ffmpegInterop. Unfortunately as I read, WebRTC needs at least some kind of signaling server which manages the connection between the clients. ffmpegInterop seems to be really difficult to integrate into unity.

  


  I am completely new to the topic of low latency video streaming and a bit lost in this complex environment. Can anybody give me an advice about a promising and extendable way to receive a low latency video stream ?

  


• bad audio mic recording quality with ffmpeg compared to sox

  1er juillet 2021, par user2355330

  I am contacting you as after 3 days of searching I am stuck on a really simple point.

  


  I want to record the sound of my mic on MacOS using ffmpeg.

  


  I managed to do it using the following command :

  


  ffmpeg -f avfoundation -audio_device_index 2 -i "none:-" -c:a pcm_s32l alexspeaking.wav -y -loglevel debug


  


  The issue is that each time I am speaking, there are cracks and pop in the sound...

  


  I tried to use sox and it gave me a perfect and crystal clear sound and I have no idea why... Below is the output of the sox command :

  


  sox -t coreaudio "G935 Gaming Headset" toto.wav -V6
sox:      SoX v
time:     Nov 15 2020 01:06:02
uname:    Darwin MacBook-Pro.local 20.5.0 Darwin Kernel Version 20.5.0: Sat May  8 05:10:33 PDT 2021; root:xnu-7195.121.3~9/RELEASE_X86_64 x86_64
compiler: gcc Apple LLVM 12.0.0 (clang-1200.0.32.27)
arch:     1288 48 88 L
sox INFO coreaudio: Found Audio Device "DELL U2721DE"
sox INFO coreaudio: Found Audio Device "G935 Gaming "
sox DBUG coreaudio: audio device did not accept 2 channels. Use 1 channels instead.
sox DBUG coreaudio: audio device did not accept 44100 sample rate. Use 48000 instead.
Input File     : 'G935 Gaming Headset' (coreaudio)
Channels       : 1
Sample Rate    : 48000
Precision      : 32-bit
Sample Encoding: 32-bit Signed Integer PCM
Endian Type    : little
Reverse Nibbles: no
Reverse Bits   : no
sox INFO sox: Overwriting `toto.wav'
sox DBUG wav: Writing Wave file: Microsoft PCM format, 1 channel, 48000 samp/sec
sox DBUG wav:         192000 byte/sec, 4 block align, 32 bits/samp
Output File    : 'toto.wav'
Channels       : 1
Sample Rate    : 48000
Precision      : 32-bit
Sample Encoding: 32-bit Signed Integer PCM
Endian Type    : little
Reverse Nibbles: no
Reverse Bits   : no
Comment        : 'Processed by SoX'
sox DBUG effects: sox_add_effect: extending effects table, new size = 8
sox INFO sox: effects chain: input        48000Hz  1 channels (multi) 32 bits unknown length
sox INFO sox: effects chain: output       48000Hz  1 channels (multi) 32 bits unknown length
sox DBUG sox: start-up time = 0.051332
In:0.00% 00:00:07.13 [00:00:00.00] Out:340k  [      |      ]        Clip:0    ^C
sox DBUG input: output buffer still held 2048 samples; dropped.
Aborted.
sox DBUG wav: Finished writing Wave file, 1359872 data bytes 339968 samples


  


  I am pretty sure the issue is linked to the way the encoding is done and the params I used with ffmpeg but I don't seem to be able to grasp which one I must use.

  


  Any ideas if there are ffmpeg experts here ?